Voice optimization in a network having voice over internet protocol communication devices

ABSTRACT

An apparatus and a method of optimizing voice quality on a network having end-points that are voice over Internet Protocol (IP) devices. Default parameters of the end-points are initialized. Network performance parameters are measured and evaluated to determine whether they signify that connection to the network is below a desired level of operation. If so, the default parameters of the end-points are re-set based on the evaluation. The adjustment may entail re-negotiating the CODEC connection and re-setting parameters for packet size and resetting parameters for jitter buffer size.

CROSS-REFERENCE TO RELATED PATENT APPLICATIONS

The present application is a continuation of pending U.S. patentapplication Ser. No. 09/750,766, filed Dec. 28, 2000, entitled “VOICEOPTIMIZATION IN A NETWORK HAVING VOICE OVER THE INTERNET PROTOCOLCOMMUNICATION DEVICES.” The present application is also related to U.S.patent application Ser. No. 09/220,232, filed Dec. 23, 1998, issued asU.S. Pat. No. 6,560,223, on May 6, 2003.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The invention relates to optimizing voice quality on networks that havevoice over Internet Protocol communication 30 devices. The optimizationis based on taking and evaluating measurements of network performancepertaining to quality of the network connection.

2. Discussion of Related Art

The contents of U.S. Ser. No. 09/220,232, now U.S. Pat. No. 6,560,223,are incorporated herein by reference.

Support for end-to-end voice calls using the Internet as an alternativeto traditional public switched telephone networks (PSTN) isconventional. Unlike the PSTN, which is circuit-switched, the Internetis packet-switched; communication on the Internet is accomplished bytransmitting and receiving packets of data.

In addition to data, each packet contains an address to ensure that itis routed correctly. The format of these packets is defined by theInternet Protocol (IP). One type of allowable data is encoded, digitizedvoice. Thus, voice over IP (VOIP) is voice that is packetized as definedby IP, and communicated over the Internet or some other IP network fortelephone-like communication.

Devices such as personal computers (PC) that are VoIP ready, wirelessVoIP devices, intelligent phones that are VoIP ready, conferencingunits, trunks that are VoIP ready, multi media devices that are VoIPready, and plain old telephone system (POTS) telephones connected toVoIP gateways may be VoIP endpoints. Examples include Microsoft'sNetMeeting, Nortel Networks's soft-client and IP based personal digitalassistant (PDA) devices.

These end-points require real-time performance over IP networks. The endpoints are assumed to be distributed on one or more IP networks. Somedevices will be on the same subnet on the same local area network. (LAN)segment while others will be on different subnets, some of which couldbe at the end of a slow wide area network (WAN) connection. A problemarises, however, with respect to voice quality and other quality ofservice (QoS) parameters in an IP network.

Conventionally, bandwidth is reserved for particular IP end points bymarking their packets with a priority level. However, about 90% of thenetworks today are not policy managed. With respect to the 10% that arepolicy managed, there are no solutions to the problems that exist whenthe bandwidth is insufficient to support a badly configured IPend-point.

Conventional software tools are available to help assess the performanceof a network by measuring parameters of network performance. Such toolsinclude a Ping tool, a Network trace tool and a packet loss measurementtool. These tools are standard with MICROSOFT WINDOWS product. They runfrom the command line (found under Start, Run on all WINDOWS operatingsystems). The commands are ‘ping IP_Address’ or ‘Trace IP_Address’.These tools are available with TCP/IP protocol stacks found underoperating systems such as Linux/VxWorks, etc.

With the telecommunications networks transitioning from circuitswitching to packet switching, phones, trunks and wireless devices areexpected to be VoIP capable. Controlling these phones, trunks, andwireless devices to optimize media performance is desired.

BRIEF SUMMARY OF THE INVENTION

An aspect of the invention resides in an apparatus and a method ofoptimizing voice quality on a network having endpoint devices, such asvoice over Internet Protocol (IP) devices. The invention includesinitializing default parameters for the end-point devices with respectto choice of preferred CODEC, number of voice samples per IP packet andjitter buffer size. Further, performance parameters are measured and ifwarranted by an indication from the performance parameters that theconnection to the network is below a desired level of operation, thedefault parameters are adjusted. The measurements include networkbandwidth, number of network hops to the IP end-point devices, roundtrip delay and packet loss over a time duration. Software tools used toobtain the measurements may include a ping tool, a network trace tooland a packet loss measurement tool. The adjustment of the defaultparameters may involve re-negotiating a CODEC connection, re-setting thepacket size and/or re-setting the jitter buffer size.

Preferably, a private branch exchange is on the network. A register isconfigured to register the end-point VOIP devices with the privatebranch exchange (PBX) on the network. In response to the registercompleting registration of the end-point devices with the PBX, aterminal proxy server may set the default parameters of the registeredend-point devices on the basis of measured performance parameters.Thereafter, the default parameters are adjusted as necessary to betteroptimize the connection on the basis of changes to the performanceparameters as measured by the software tools.

BRIEF DESCRIPTION OF THE SEVERAL VIEWS OF THE DRAWING

For a better understanding of the present invention, reference is madeto the following description and accompanying drawing, while the scopeof the invention is set forth in the appended claims.

FIG. 1 shows a flow chart representing a condition of the invention atinitialization time.

FIG. 2 shows a flow chart representing a three-phase approach inaccordance with the invention.

FIG. 3 shows a schematic representation of IP telephony in accordancewith the invention.

DETAILED DESCRIPTION OF THE INVENTION

Before discussing the FIGS. 1-3, a brief discussion about various termsused in this patent application may be helpful in better understandingthe invention.

Bandwidth usually refers to maximum available bit rate for a specificapplication. In a specific embodiment, synchronous, interactive, andreal-time applications, which are bandwidth-sensitive, can requireminimum bandwidth guarantees, and can require sustained and burst-scalebit-rates. On the other hand, network administrators may want to limitbandwidth taken by non-productive traffic. However, even thoughbandwidth may be allocated for specified applications, it does not meanthat these applications may be using that bandwidth.

Jitter generally refers to variation in delay (e.g., the delay is notconstant for all packets of a given flow) for a particular application.Real-time applications have maximum jitter limits. Applications such asreal-audio and video do some advanced buffering in an attempt toovercome variation in packet delays—the amount of buffering isdetermined by the expected jitter.

A jitter buffer is a variable length buffer that stores voice packets.The size of the jitter buffer may be dependent upon the variation in theend-to-end packet delay in the network and can be dynamically adjustedbased upon, for example, this packet delay variation.

Latency generally refers to the delay experienced by a packet from thesource to destination. Latency requirements are typically specified asmean-delay and worst case delay in some cases. Delay is a result ofpropagation delay, due to physical medium and queuing at intermediatenodes such as routers, gateways, servers, etc. A certain portion of thedelay can be controlled by changing the way in which the queues areserviced at the intermediate nodes, and by controlling congestion atbottleneck points. Some examples of delay measurements are packetround-trip delay and connection response time.

A Ping tool includes a routine that sends a packet onto the network andobtains a value of the average delay encountered by that packet inreaching the destination and returning.

A network trace tool includes a routine that captures and records eventsand states that occur while the network is operating. Events in a tracechronology may be paired, forming event-pairs. Examples of event-pairsinclude the start and end times.

A packet loss measurement tool measures the rate of packet loss over aperiod of time. Packet loss is a rate of loss in a packet or a portionof packets that is generally caused by noise or failure of networkelements (e.g., routers, servers) to forward or deliver packets. Packetloss is usually an indication of severe congestion, overload of anelement, or element failure (e.g., if a server is down). Even if thepacket was not dropped but just delayed, protocols and applications mayassume it was lost. Packet loss can cause application timeouts, loss ofquality or delay due to retransmitted packets. Indirect results ofpacket loss may also be measured (e.g., connection retries or dataretransmits).

The default parameters of VoIP end-point devices need to be initialized(FIG. 1) and adjusted later as necessary (FIG. 2). FIG. 3 illustrates anexample of VoIP end-point devices on a LAN. These devices include awired VoIP Client (Set B), a wired soft VoIP Client (Set C) and WirelessVoIP CLIENTS (SETS D & E). These identified end-point devices are forillustration purposes only, there can be more or less end-point devicesof different types and additional LANs and additional subnets, etc.

The LAN is connected to a WAN via a call server, terminal proxy serverand router in a conventional manner. A wired VoIP Client (Set A) is incommunication with one or more of the VoIP devices of the LAN via theWAN. The parameters that may 25 be optimized include CODEC selection,packet size (that is, the number of frames per packet), desired latency,packet loss, available bandwidth, number of router hops, and jitterbuffer size.

When a VoIP end-point device registers with an IP PBX, the IP PBX mayperform a number of tests to determine the optimum configuration forthat end-point. It can measure network bandwidth, the number of networkhops (if any) to the end point, the round trip delay and/or the packetloss over a short period of time. These tests can easily be done using aping tool, a network trace tool and/or a packet loss measurement tool.After initialization, one or more of these tools can be used on regularperiods to determine network performance. During active communications,a call re-establishment can be initiated (manually or automatically) tore-apply these parameters.

Turning to FIG. 1, at the time of initialization, the VoIP end-pointdevice registers with the PBX. In response to the registration, aterminal proxy server uses a protocol to instruct the end-point deviceto use a CODEC of a particular type, a jitter buffer of a particularsize, a frame size of so many voice samples, etc. That is, the terminalproxy server sets the default settings for that VoIP end-point device.When the VoIP end-point device attempts to make a call, the subnetaddresses of different IP sets are compared.

If the comparison shows that the subnet addresses differ, then thedefault is set using G729A as the preferred CODEC type for both IPend-point devices, although other CODECS could be chosen as the defaultCODEC. Otherwise, if the comparison shows that the subnet addresses arethe same, then the default is set using G711 as the preferred CODEC typefor both IP sets, although other CODECS could be chosen as the defaultCODEC.

In addition, software tools are used to measure network performanceparameters (e.g., jitter, available bandwidth, delay, packet loss,latency, etc). In obtaining such measurements, a ping tool, a networktrace tool and/or a packet loss measurement tool may be used to obtainthe necessary information for selection of the default settings.

The Ping tool measures latency and bandwidth of the network between IPend-point devices and/or between end-point devices and the PBX. It maydo so by measuring the round trip delay between an IP end-point deviceand a test point. Bandwidth is measured by measuring the throughput persecond of a communication link. In cases where a voice path is directlybetween two IP end-point devices and not through an intermediarygateway, the test point requests the IP end-point devices to make themeasurement on its behalf with respect to latency and bandwidth betweenthe two IP end-point devices that are connected together. The testresults become known to the terminal proxy server, which makes thedecisions on these parameters on behalf of the IP end-point devices.

If the latency is low and the bandwidth is high, Table 2 is used toselect a small number (about 1-4 frames by example only) of voice framesper packet. If the latency is high and the bandwidth is low, Table 1 isused to select a larger number (about 7-10 frames by example only) ofvoice frames per packet for optimum efficiency.

If the bandwidth is low, such as at 64 kbs, a CODEC with a highcompression ratio should be used (as exemplified by G723.1) and thelatency will always increase as a consequence. In general, the lessbandwidth, the worse the latency will likely be. On the other hand, ifthe bandwidth is high, the latency could also be high if the CODECselection, packet size and jitter buffer size is chosen poorly. Byrecognizing these conditions, however, appropriate changes may be madeto these parameters to optimize latency and voice quality.

If there is spare bandwidth, then network efficiency becomes less of anissue. As a result, stuffing a small number, such as less than three,frames per packet produces less delay/latency and yields better voicequality.

The Network trace tool is used to determine the number of Router Hopsbetween IP end-point devices and/or between the end-point devices andthe PBX. If the number of hops is small (1-2 for example) and thebandwidth is high, a small number of frames per packet is used fromTable 2 for G711. Otherwise, if the number of hops is large and thebandwidth is small, a large number of frames per packet is selected fromTable 1 for G729A.

FIG. 2 illustrates a three-phase approach to optimize a VoIP connection,namely, initializing, network performance monitoring and dynamicallyintervening or correcting. The initializing phase has been discussed indetail with respect to FIG. 1 and involves optimizing the configurationof the end point devices. The end point is a frame of reference and canbe regarded as either the terminal or the gateway.

Network performance monitoring involves periodically performing (e.g.;every few minutes, or other time intervals), a maintenance task thatmeasures the network performance. The maintenance task may involve theuse of the Ping tool, the Network trace tool and/or the data packet lossmeasurement tool. If the network is found to have changed significantlyas compared to its condition at the time of initialization, the stepsmentioned for Initialization with respect to FIG. 1 may be repeated tore-optimize the network. This could be done automatically or at therequest of an end-point device.

The default parameters for the end-point devices should be changed wherethe measured performance parameters signify that connection to thenetwork is below a desired level of operation. If the latency exceeds200 milliseconds, for instance, changes should be made to the defaultparameters. For instance, changing the default parameters for the packetsize and CODEC type will help. If the bandwidth is below a thresholdlevel of 128 kbs, the default parameters for the preferred CODEC typeneeds to be changed. If the packet loss is more than 5%, the defaultparameters should be changed to a low number of voice samples per frame.

The terminal proxy server and the end-point devices make the necessarymeasurements with the software tools and report the findings back to theterminal proxy server, which uses conventional protocol to instruct theend-point devices to make the appropriate changes in their defaultparameter settings. Conventional programming is used to implement theinstructions from the terminal proxy server.

The next phase involves dynamic intervention or correction when the endpoint devices are active on a session. For example, during an activesession, when two end-point devices are in voice communication,corrective measures may need to be taken to improve voice quality. Usingintelligence in the IP sets, an indication of the voice quality isreported to a centralized terminal proxy server (see FIG. 3).

If the voice quality at either end-point device has degraded below a setthreshold value, then the steps mentioned in connection with FIG. 1 atthe time of initialization may be automatically implemented. Suchdegradation signifies that conditions of the network may have changedsignificantly as compared to the time of initialization. As the voicechannel is re-optimizing, a slight interruption may occur in the voicepath. The voice paths may be muted by the end points while thistransition is taking place. This would make the transition seamless froman end user perspective. Preferably, the user is prompted to acknowledgethat such re-optimization action is to be taken, although such promptingis hot required.

At the time of initialization, an IP-PBX may be used that could setcertain parameters for each device. The parameters are the preferredCODEC, the number of Voice Samples per IP packet, and the jitter buffersize. For VoIP end-point devices that are installed on the same subnet,the IP^PBX may set the parameters the same as follows. The CODECselection is G711, the number of voice samples that would be sent witheach packet is legs than five (and thus small, yielding approximately50% overhead in the TCP/IP frame), and the jitter buffer selected wouldbe very small or none.

This three-phase approach will give the best audio quality with thelowest amount of latency and will use the minimum amount of digitalsignal processor (DSP) resources. This approach, however, does notensure a good connection, because it only optimizes the availableconnection. This approach works well if there is sufficient bandwidth onthe local subnet and if the packet loss is small (this is very probablein this scenario).

For devices on a different Subnet, the IP-PBX may perform a number oftests before defaulting the system with the above parameters. However,to save processing time and power, it could default the system to theabove parameters and then perform the tests to adjust the defaultsettings as necessary based on results from the tests.

By pinging the end-points with data, it is possible to measure thethroughput, latency and packet loss of the network connection. It islikely that a G729A CODEC, a much larger packet size (e.g., voicesample) and a sizable jitter buffer to allow for out of order packetswould be selected.

By continually monitoring the network it is possible to dynamicallychange the default parameters associated with a VoIP end point so thatwhen a new call is started or received the network connection isoptimized.

If a network is experiencing performance issues during a VoIP call, theIP-PBX could either automatically re-negotiate the CODEC connection andre-set the parameters for packet size and/or jitter buffer. The optionto perform such dynamic intervention during media transmission couldalso be provided to the user with a manual interface to start thisactivity (e.g., similar to the “scan” button on a cordless phone).

Since the bandwidth of the spectrum is much more limited for wirelessdevices than for that of a wired LAN, the default parameters could beset based on whether the WLAN was a direct sequence spread spectrum(DSSS) (such as 11 mbs) or a frequency hopping (FH) spread spectrum(such as 2 mbs) or other type of spread spectrum. DSSS and FH spreadspectrum are two types of modulation techniques for use in wirelesscommunication systems. Depending upon system requirements, eachmodulation has its set of advantages and disadvantages. In a messagingenvironment, frequency hopping is more attractive than direct sequencebecause it requires no power control at a portable subscriber unit orend-point device. DS is more attractive for location findingapplications (using time of arrival) or where spectral reconstructioncan be used in interference cancellation.

As a consequence of following this three-phase approach, voice qualitybecomes optimized for those IP networks that have not been provisionedfor quality of service (QoS) and is further optimized if the networkemploys QoS measures. This is a self-learning and self-correctingsolution to maximize voice and other media performance in an IP network.

While the foregoing description and drawings represent the preferredembodiments of the present invention, it will be understood that variouschanges and modifications may be made without departing from the spiritand scope of the present invention.

What is claimed is:
 1. A method of voice performance improvement in apacket switched network, comprising: initializing an end-point device onthe network, wherein the initializing comprises setting initialparameters affecting VoIP operation of the end-point device andperforming at least one test to determine a configuration for theend-point device corresponding to a desired level of operation of aconnection to the network by the end-point device; measuring at leastone performance parameter of the network external to the end-pointdevice; and evaluating whether the measured performance parameterindicates that the connection to the network is below the desired levelof operation and, if so, transmitting a signal indicating that theconnection to the network is below the desired level of operation. 2.The method of claim 1, wherein the step of initializing an end-pointdevice comprises setting at least one initial parameter for theend-point device comprising one or more of: a choice of a preferredCODEC; a number of voice samples per packet; and a jitter buffer size.3. The method of claim 1, further comprising responding to the signalindicating that the connection to the network is below the desired levelof operation by adjusting at least one parameter of the end-pointdevice.
 4. The method of claim 3, wherein the adjusting comprisesperforming at least one function selected from a group consisting of:renegotiating a CODEC connection; re-setting packet size parameters; andre-setting a jitter buffer.
 5. The method of claim 3, wherein the stepof adjusting at least one parameter is initiated by a user.
 6. Themethod of claim 3, wherein the adjusting step is carried out duringtransmission of media to the end-point device.
 7. The method of claim 1,wherein the step of measuring at least one performance parametercomprises measuring at least one performance parameter selected from agroup consisting of: throughput; latency; packet loss; bandwidth; numberof network hops to at least one end-point device; and round trip delay.8. The method of claim 1, wherein the step of measuring at least oneperformance parameter comprises measuring performance with at least onetool selected from a group consisting of: a ping tool; a network tracetool; and a packet loss measurement tool.
 9. The method of claim 1,further comprising registering the end-point device with a PrivateBranch Exchange (PBX) on the network.
 10. The method of claim 9, whereinthe measuring step comprises measuring with the PBX at least oneperformance parameter characterizing the network between the PBX and theend-point device.
 11. An apparatus to effect voice performanceimprovement in a packet switched network, comprising: an initializerconfigured to initialize at least one initial parameter for an end-pointdevice on the network and to perform at least one test to determine aconfiguration for the end-point device corresponding to a desired levelof operation of a connection to the network by the end-point device; ameasurer configured to measure at least one performance parameter of thenetwork external to the end-point device; an evaluator configured tomake a determination as to whether the at least one measured performanceparameter indicates that the connection to the network is below thedesired level of operation; and a transmitter configured to transmit asignal indicating that the connection to the network is below thedesired level of operation.
 12. The apparatus of claim 11, wherein theinitializer is configured to set at least one initial parameter for theend-point device comprising one or more of: a choice of a preferredCODEC; a number of voice samples per packet; and a jitter buffer size.13. The apparatus of claim 11, further comprising an adjuster responsiveto the signal indicating that the connection to the network is below thedesired level of operation to adjust the at least one performanceparameter of the end-point device.
 14. The apparatus of claim 13,wherein the adjuster is configured to perform at least one functionselected from a group consisting of: renegotiating a CODEC connection;re-setting packet size parameters; and re-setting a jitter buffer. 15.The apparatus of claim 13, wherein the adjuster is configured to adjustat least one parameter when initiated by a user.
 16. The apparatus ofclaim 13, wherein the adjuster is configured to carry out adjustmentsduring transmission of media to the end-point device.
 17. The apparatusof claim 11, wherein the measurer is configured to measure at least oneperformance parameter selected from a group consisting of: throughput;latency; packet loss; bandwidth; number of network hops to at least oneend-point device; and round trip delay.
 18. The apparatus of claim 11,wherein the measurer is configured to measure performance with at leastone tool selected from a group consisting of: a ping tool; a networktrace tool; and a packet loss measurement tool.
 19. The apparatus ofclaim 11, further comprising a registration entity configured toregister the end-point device with a Private Branch Exchange (PBX) onthe network.
 20. The apparatus of claim 19, wherein the PBX isconfigured to measure the at least one performance parametercharacterizing the network between the PBX and the end-point device.